June 2012 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Are you using NDLB-force-rport = "safe" ?
by Michael Collins 25 Jun '12

25 Jun '12

Hi gang, A friend of mine asked how many people are using this. Are you? It's mostly for Polycoms and non-polys to have force rport in a typical hosted environment. If you have been using this setting in production please give us some feedback. Thanks! -Michael

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[Freeswitch-dev] bandtel SIP TRUNK config
by victor.velo＠videotron.ca 21 Jun '12

21 Jun '12

Hello, I need some help to validate my bandtel sip trunking config... I get the following info from bandtel: Your SIP User Name: 201XXXXXXX Your SIP Password: XXXXXX Your ANI: 207XXXXXX Your Toll Free/s: N/A Your DID/s: 207XXXXX (prefix with0317) --- DNS Server Addresses: >65.175.129.149 >66.237.65.90 Registration must be sent to: registrar.bandtel.com •Outbound Calls: Outbound calls must be sent to the following proxy: Primary: proxy1.bandtel.com; Secondary: proxy2.bandtel.com my setting: ....... cat /etc/resolv.conf nameserver 65.175.129.149 nameserver 66.237.65.90 .................. cat 01_bandtel.xml <include> <gateway name="bandtel.com"> <param name="username" value="201XXXXXX"/> <param name="password" value="XXXXX"/> <param name="from-user" value="+1207XXXXXXX"/> <param name="proxy" value="registrar.bandtel.com"/> <param name="outbound-proxy" value="proxy1.bandtel.com"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="caller-id-in-from" value="true"/> </gateway> </include> ....................... cat 01_bandtel_did.xml <include> <extension name="bandtel_did"> <condition field="destination_number" expression="^(207XXXXX)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="XXXX XML default"/> </condition> </extension> </include> ............. I don't get any error in the log file and I don't get anything by calling the bandtel DID.... Do I miss some settings or params somewhere???? Thank for the helps, Victor Velo

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[Freeswitch-dev] New wiki page: Language_Files
by Michael Collins 21 Jun '12

21 Jun '12

Hello all, I've created a new wiki page: http://wiki.freeswitch.org/wiki/Language_Files I've wanted this for some time and Brian's OCD finally took over and I just *had* to create this page. :) I've added languages that I know about or have heard about. *This page is not complete, and it needs your help.* If you know anything about any language files then please please go to that page and add what you know. If you have any issues with editing the wiki then let me know and I will be happy to assist. In fact, if need be I will do a 20-minute tutorial on wiki editing on an upcoming community conference call. Thanks! -Michael

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[Freeswitch-dev] FreeSWITCH Conf Call Today: SIP 101
by Michael Collins 20 Jun '12

20 Jun '12

Hey all, Don't forget that at 1pm EDT, 10am PDT we have the FreeSWITCH community conference call. Darren Schreiber will be doing the SIP 101 discussion. We look forward to having you all with us. -Michael

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[Freeswitch-dev] sendevent NOTIFY with uuid - the tag from RINGING response is not copied to NOTIFY
by Jon Bruel 19 Jun '12

19 Jun '12

I have tested how I can control a SIP-phone receiving a call to go off-hook. This is possible by sending a NOTIFY with the SIP Event-header set to talk. My problem is that the NOTIFY message does not include exactly the same SIP To-header information as the FS received as a RINGING response to the INVITE: The tag from the RINGING response, which the phone adds to the SIP To-header, is not included in the NOTIFY. Snom-phones - and possibly other SIP phones as well, uses this tag in the SIP To-header to pinpoint the exact call to be answered. The information is actually present before the NOTIFY is sent. I have tested it in the dialplan using the info application just before firing off the sendevent NOTIFY. Is there any reason for this behaviour, which I should be aware of before dabbling with the code? I have checked the mod_sofia.c code and have found the code which sends the NOTIFY (with the uuid, event and message type as input parameters): if (uuid && ct && es) { switch_core_session_t *session; private_object_t *tech_pvt; //Jon changed: SIPTAG_SUBSCRIPTION_STATE_STR("terminated;reason=noresource") to SIPTAG_SUBSCRIPTION_STATE_STR("active") if ((session = switch_core_session_locate(uuid))) { if ((tech_pvt = switch_core_session_get_private(session))) { nua_notify(tech_pvt->nh, NUTAG_NEWSUB(1), SIPTAG_SUBSCRIPTION_STATE_STR("active"), SIPTAG_EVENT_STR(es), SIPTAG_CONTENT_TYPE_STR(ct), TAG_IF(!zstr(body), SIPTAG_PAYLOAD_STR(body)), TAG_END()); } switch_core_session_rwunlock(session); } } But in order to change the behaviour, I may need to look somewhere else. Some hints as to where to look would be appreciated! Regards Jon

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 18 Jun '12

18 Jun '12

We hope you're having a nice Monday today. The past week was a busy one both for the FreeSWITCH<http://www.freeswitch.org?cc12-0618>project and for ClueCon <http://www.cluecon.com?cc12-0618>. Last week's community conference call was a nice discussion about some of the things that will be happening at ClueCon this year. We also had a visit from Diana Cionoiu of the YATE <http://yate.null.ro/pmwiki?cc12-0618> project. This coming Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books and co-founder of the 2600hz project <http://www.2600hz.org?cc12-0618>, on our call to give his world-famous "SIP 101" presentation. We definitely look forward to that discussion. We also wanted to highlight an interesting blog post<http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html?cc…>by our friend and community member Kristian Kielhofner. Its provocative title is: "Everything you wish you didn't need to know about VoIP." Over the years Kristian has collected a lot of knowledge about interoperability - and lack thereof - between various VoIP devices and servers. Whether you're a VoIP novice or veteran we think you will appreciate seeing this knowledge written down for the benefit of all. In ClueCon news are happy to announce two training session and a birds of a feather (BOF) meetup for this year's event. Paid attendees of ClueCon will have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH training will be conducted by the aforementioned Darren Schreiber. The OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu, lead developer of the OpenSIPS project. The training sessions will take place on Monday, August 6. On the evening of Wednesday, August 8 we will be having a VoIP security ("VoIPSec") meetup for those interested in discussing this subject with others who share an interest. Among those who will be present are Phil Zimmermann of PGP and ZRTP<http://zfone.com?cc12-0618>fame. Be sure to register right away so that you can get your extra chances to win. Anyone who registers before July 4th of this year will receive 10 entries in the great ClueCon giveaway. We have a lot of interesting items to give away this year, so stay tuned. And while you are at the ClueCon site to register <http://www.cluecon.com/register?cc12-0618>, but sure to review the schedule <http://www.cluecon.com/schedule?cc12-0618> as we've added a number of new speakers in the past week. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0618

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 11 Jun '12

11 Jun '12

Happy Monday to you all! Last Wednesday we had a nice discussion on the weekly conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls?cc12-0611>about mod_httapi <http://wiki.freeswitch.org/wiki/Mod_httapi?cc12-0611>. It turns out that there is a fair amount of interest in the great module but that there are some questions about how to handle certain scenarios, such as "session tracking." Thanks to Raymond Chandler for being available to answer a lot of those questions. We will be doing a followup discussion that focuses on Raymond's PHP examples for using mod_httapi. These examples do a good job of demonstrating the power and ease of HTTAPI. ClueCon <http://www.cluecon.com?cc12-0611> season is upon us and our list of speakers and sponsors continues to grow. Packt Publishing<http://packtpub.com?cc12-0611>is now on board as a media sponsor and we hope to have some nice items from them to give away at the conference. Keep checking the schedule<http://www.cluecon.com/schedule?cc12-0611>for updates as we are confirming new speakers each week. We are also happy to report that the Wyndham has completed its changeover and is officially known as the Hyatt Chicago Magnificent Mile<http://www.chicagomagnificentmile.hyatt.com?cc12-0611>. (As of this writing their web site is undergoing maintenance, so be patient.) We are excited to see what's in store at this revamped hotel and can't wait to see everyone there this August! -- Michael S Collins ClueCon Team http://www.cluecon.com?cc12-0611 877-7-4ACLUE

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[Freeswitch-dev] Andre Mendes invited you to Dropbox
by Dropbox 11 Jun '12

11 Jun '12

Andre Mendes wants you to try Dropbox! Dropbox lets you bring all your photos, docs and videos with you anywhere and share them easily. Get started here: http://www.dropbox.com/link/20.N7kjcPs2RS/NjIxOTQ4NDg5Nzc?src=referrals_bul… - The Dropbox Team ____________________________________________________ To stop receiving invites from Dropbox, please go to http://www.dropbox.com/bl/074747b8b714/freeswitch-dev%40lists.freeswitch.org

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[Freeswitch-dev] video formats, playback and recording features
by Seven Du 08 Jun '12

08 Jun '12

Hi there, As we have mod_fsv, mod_mp4v, mod_vlc, mod_dingaling, mod_rtmp, mod_rtsp, some already support video and some are possible. In addition to p2p videos, I'd like to also get video playback, recording, eavesdroping working. So I did some research and here's some quetions: Basically there's two ways to support video 1) formats. formats support a large range of audios, but not videos. So it is possible to extend to add something like switch_file_read_video/switch_file_write_video etc. And also extend playback, record eavesdrop, fifo, esf_page_group ... to support video, and also need to extend video media_bug ... Video files like mp4 also including audio tracks 2) as the current solutions in modules on the top, provide different tools like play_fsv, play_mp4, etc to add video support The 2) perhaps natural as it goes, and 1) perhaps more elegant but may add complexity and possible resource/performance problems to pure audio usage? Advice ? Is there any interest to support video transcodings ? btw, any working in progress of webrtc in FS ? I saw this http://code.google.com/p/sipml5/ sounds cool. Thanks, Seven. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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[Freeswitch-dev] FreeSWITCH Conf Call Is On!
by Michael Collins 06 Jun '12

06 Jun '12

Come join us! http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06 We're going to talk about mod_httapi and a few other fun things. Thanks! -Michael

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